1. Field of Invention
The present invention relates generally to the field of communications. More specifically, the present invention is related to a system and method for transferring TDM data over packet switched networks, such as IP networks.
2. Discussion of Prior Art
T-1 (DS1) trunks are circuit switched data networks supporting data rates of 1.544 Mbits per second. A T-1 trunk can carry 24 individual 64 Kbits per second channels, each of which may carry data or telephony quality voice. Similarly, E1 trunks are circuit switched data networks supporting data rates of 2.048 Mbps (32 channels at 64 Kbps). T-3 and E3 trunks support data rates of 44,736 and 34,368 Kbps, respectively. Together T1, E1, T3, E3 and similar circuit switched serial networks are known as TDM networks.
TDM, short for Time Division Multiplexing, is a type of multiplexing that combines data streams by assigning each stream a different time slot in a set. TDM repeatedly transmits a fixed sequence of time slots over a single transmission channel. Within T-Carrier systems, such as T-1 and T-3 (DS3), TDM combines Pulse Code Modulated (PCM) streams created for each in conversation or data stream.
ATM, short for Asynchronous Transfer Mode, represents a network technology based on transferring data in cells of a fixed size. The cell used with ATM is relatively small (53 bytes) compared to units used with older technologies. The small, constant cell size allows ATM equipment to transmit video, audio, and computer data over the same network, and assure that no single type of data hogs the line.
Current implementations of ATM support data transfer rates of from 1.544 (T1) to 622 Mbps (megabits per second). This compares to a maximum of 1000 Mbps (GbETH) for Ethernet, the current technology used for most LANs. ATM over IP pseudo OSI layers comprise an upper protocol, ATM Service Specific Convergence Sublayer (ATM-SSCS), necessary to translate between the ATM layer and RTD/UDP/IP sublayers. The User Datagram Protocol (UDP) is a connectionless protocol that, like TCP, runs on top of IP networks. Unlike TCP/IP, UDP/IP provides very few error recovery services, offering instead a direct way to send and receive data grams over an IP network. Sub-layers provide the Ethernet type, MAC header and PHY Ethernet respectively.
High-speed IP-based networks are the latest innovation in the world of communications. The capacity of these networks is increasing at a prodigious rate, fueled by the popularity of the Internet and decreasing costs associated with the technology. Worldwide data traffic volume has already surpassed that of the telephone network, and for many applications, the pricing of IP traffic has dropped below the tariffs associated with traditional TDM service. For this reason, significant effort is being expended on VoIP technologies. For users who have free, or fixed-price Internet access, Internet telephony software essentially provides free telephone calls anywhere in the world. To date, however, Internet telephony does not offer the same quality of telephone service as direct telephone connections. There are many Internet telephony applications available. Some come bundled with popular Web browsers; others are stand-alone products. Internet telephony products are sometimes called IP telephony, Voice over the Internet (VOI) or Voice over IP (VOIP) products.
Inherent in all forms of VoIP is revolutionary change, whereby much of the existing telephony infrastructure will be replaced by novel IP-based mechanisms. Despite the hype, this effort has been more protracted and less successful than initially expected. Today""s telephony technology, both those portions that VoIP aims to replace and those to which VoIP must interface, is extremely complex. Revolutionary implementations of its hundreds of features and thousands of variations cannot be expected to be developed in a short time frame.
The present communications revolution has been focused on the Internet and the Internet protocol (IP). The prior art, however, has failed to teach a viable solution to handling TDM over Internet Protocol (IP). In addition, the use of TDM over IP for Voice over IP has not been heretofore possible.
Why Use IP Networks? The existing telephony infrastructure has an extremely high reliability (99.999%), supports reasonable audio quality (Mean Opinion Score, or MOS, 4.0 on a scale of 1 to 5), has almost universal market penetration, and offers a rich feature set. Accordingly, extremely potent incentives are required before one should consider supplanting it. There are two such incentives, one economic and one technological.
The part of the economic advantage of IP networks is shared by all packet networks; namely, that multiple packetized data streams can share a circuit, while a TDM timeslot occupies a dedicated circuit for the call""s duration. Under xe2x80x9cpolite conversationxe2x80x9d assumption of each party speaking only half of the time, and the xe2x80x9coptimal engineeringxe2x80x9d assumption of minimal overhead, packet networks will, on average, double the bandwidth efficiency, thus halving operational costs. Taking overhead and peak statistics into account, the savings will be somewhat less, but a 30% reduction is attainable. However, it is doubtful whether this savings alone would be a strong enough encouragement to make the switch from TDM to IP.
The added catalyst has to do with the raw rates for data traffic as compared to voice traffic. At present, data communications are metered separately from traditional voice communications and are offered at substantial savings. These savings are partly due to tariffs and access charges that increase the cost of traditional voice services, and partly due to the attractive pricing of IP traffic. Put another way, voice service pricing is still mostly determined by incumbent carriers with high overhead costs, while IP traffic costs are much more competitive, as the provider incurs lower costs and is more focused on increasing market share.
The technological incentive has come to be called convergence. The reasoning is that technological simplification and synergy will result from consolidation of the various sources into an integrated environment. For example, a single residential information source provisioned for telephony, IP data and entertainment programming would in principle decrease end user prices, result in a single unified billing package, and eventually enable advanced services, such as video-on-demand.
The Limitations of VoIP
In principle, it would not seem difficult to carry voice over IP networks; a digitized voice signal is simply data and can be carried by a packet network just like any other data. The major technological achievement of the telephone network, that of least cost routing, has its counterpart in IP networks as well. There are, however, two fundamental problems that have to be solved before VoIP can be realistically considered to compete with TDM networks; namely, QoS and signaling.
Quality of Service
The meaning of Quality of Service is completely different for data and voice. Although most data can withstand relatively significant delay, low delay and proper time ordering of the signal are critical for voice applications, even though loss of a few milliseconds of signal is usually not noticeable. These requirements are completely at odds with the basic principles of IP networks (although not necessarily with those of other packet networks). To overcome these constraints, mechanisms such as tunneling and jitter buffers need to be employed. Additional components of voice quality such as echo cancellation and voice compression are not inherent in data-based networks at all, and need to be added ad hoc for VoIP.
Almost all of the massive RandD effort in the field of VoIP is directed towards solving these QoS problems, leaving the signaling problem largely unsolved. By signaling, we mean the exchange of information needed for a telephone call other than the speech itself Signaling consists of basic features such as the fact that the phone is off-hook or needs to ring; more advanced properties required for reaching the proper destination and billing; and still more sophisticated characteristics, such as caller identification, call forwarding, and conference calls; as well as more recent additions necessitated by intelligent networking. There are literally thousands of such telephony features, with dozens of national and local variations. Phone customers are mostly unaware of this complexity, at least until you try to deprive them of any of the features to which they have become accustomed.
Adding auxiliary information to digital voice on an IP network is in principle much simpler than signaling in telephone networks. One needn""t xe2x80x9crob bitsxe2x80x9d or dedicate CAS channels. One need only send the signaling data in some appropriate format along with the voice. Indeed, the advantage of VoIP is that it becomes possible to add features that could not exist in the classic telephony world, for example video and xe2x80x9cwhiteboards.xe2x80x9d This is true as long as the two sides to the conversation are using special VoIP terminals or computers. The problems arise when one must interface between the IP network and the standard telephony network, a connection that is imperative in light of the universal availability of standard telephone sets.
VoIP enthusiasts often emphasize conversations between two PC users or a PC user conversing with a telephone user. Consider instead a conversation between two telephone users, each connected via a standard Local Loop to a central office, but with an IP-based network replacing the TDM network between the central offices. To properly pass the requisite signaling, the IP network has to be enhanced to handle all the thousands of features and their variations (for example, 911 and *67 service). Although not an impossible task, it is one that VoIP developers have not yet accomplished.
Each of the below described references teach methods for communications using differing protocols, such as ATM over IP, across various communication standards. However, none of the references provide or suggest the present invention method of TDM over IP.
The patent to Keshav et al. (5,623,605), assigned to Lucent Technologies, Inc., provides for Methods and Systems for Interprocess Communication and Inter-Network Data Transfer. Disclosed is the transmission of data packets between source and destination devices wherein generated and received data are in ATM-formatted frames and the network transmits data in Internet protocol packets. Such data transfer is accomplished using encapsulators and decapsulators to encapsulate ATM formatted frames in data portions of IP packets for transmission on the network. (See Column 2, Line 59xe2x80x94Column 3, Line 25).
The patent to Allan et al. (5,946,313), assigned to Northern Telecom Limited, provides for a Mechanism for Multiplexing ATM AAL5 Virtual Circuits over Ethernet. This reference describes a method for encapsulating/segmenting ATM cells over Ethernet.
The patent to Aziz et al. (5,548,646), assigned to Sun Microsystems, Inc., provides for a System for Signatureless Transmission and Reception of Data Packets Between Computer Networks. Aziz et al. disclose a system for automatically encrypting (by adding an IP header) and and decrypting a data packet sent from a source host to a destination host across a network (See Column 2, Lines 13+).
The patent to Doshi et al. (5,936,965), assigned to Lucent Technologies, Inc., provides a Method and Apparatus for Transmission of Asynchronous, Synchronous, and Variable Length Mode Protocols Multiplexed over a Common Bytestream. This reference describes a system for supporting the transmission and reception of ATM over a common bytestream with a common physical layer datalink.
The patents to Watanabe (5,715,250xe2x80x94ATM-LAN connection apparatus of a small scale capable of connecting terminals of different protocol standards and A TM-LAN including the ATM-LAN connection apparatus); Acharya et al. (5,903,559xe2x80x94Method for Internet protocol switching over fast A TM cell transport), and Alexander, Jr. et al. (5,936,936xe2x80x94Redundancy mechanisms for classical Internet protocol over asynchronous transfer mode networks) provide a general teaching of IP over ATM.
The non-patent literature entitled xe2x80x9cPathbuilder S200 Voice Access Switchesxe2x80x9d http://searchpdg.adobe.com/proxies/1/70/33/43.html, provides for TDM-based voice and data traffic over frame relay or IP WAN Infrastructures.
The non-patent literature entitled xe2x80x9cProject: Gateway Applicationxe2x80x94ATM less than xe2x80x94 greater than IP,xe2x80x9d http://www.cse.ucsc.edu/xcx9crom/projects/atm_ip/atm_ip.html, describes an ATM driver over an IP driver.
The present invention offers a solution for transferring transparently E1 or T1 (or fractional E1/T1) TDM services over widely deployed high speed IP networks. This technology can be used as a migration path to Voice over IP or a complementary solution to VoIP in places where voice over IP solution is not suitable. The same TDM over IP approach can be adopted to transfer other TDM rates (e.g., E3/T3, STM1 etc.) over the IP network. Throughout the specification, claims and drawings various TDM rates may be substituted without departing from the scope of the present invention.
Whatever the precise merits, features and advantages of the above cited references, none of them achieve or fulfills the purposes of the present invention. These and other objectives are achieved by the detailed description that follows.
A method and system for processing one or more TDMs trunks for communication over IP networks, such as the Internet, includes encapsulating ATM cells (packets) using AAL1 cells within UDP over IP frames to provide synchronous bit streams into fixed size cells. This allows for an IP header to be added to the packets, with such packets forwarded to its destination host across the IP network. The destination regenerates the clock, decrypts/strips the IP header and delivers a synchronous bit stream. Furthermore, an adaptive clock is provided for clock transfer across the network. The adaptive clock regenerates the far end T1/E1 receive clock out of the incoming arrival frame rate. Frames arriving from the IP network are stored in a buffer and taken out for TDM stream assembly.